I am building a pair of speakers in my dedicated home theatre/studio. I wonder if you guys could take a look at the attached picture of the two options I am considering and tell me the best route to take.

Should I mount the mids in a line and place the tweeter beside the array or should I split up the array into two arrays and place the tweeter between the two arrays like an MMMMTMMMM configuration.

The horizontal middle of the speakers is at roughly head height when sitting in a typical couch.

The 18s will be ported with the port at the bottom of the cabinet or horn loaded with the mouth of the horn to the left side of the cabinet, placing it right in the corner of the room. (this is a left speaker shown. The right speaker will be symmetrically opposite)

The Vifa mids will be mounted with their own cabinets behind the baffle so that the pressure waves from the woofers do not have an effect on them.

Seriously, though, the horizontal polar response of this system at both the low/mid and mid/high crossover points is going to depend on the driver layout, and much more so for the HF, as the wavelengths are much smaller.

The polar response of the speaker is a direct function of the driver spacing, and the phase response of the drivers+crossover at the crossover frequency. Certain crossover designs (eg LR24) are in phase through the crossover point, while others are in phase quadrature (ie =/-90 deg), and that will change the symmetry of the polar response.

I assume that this system will be three-way triamped... You want to put the tweeter in between the mids, as that will assure symmetric polars in both the vertical and horizontal planes. Putting the driver off to the side is asking for trouble in both planes.

Now, with that question settled, what to do for the crossover points and slopes? Do you have access to a dual channel FFT analyzer, and/or an MLS analyzer? That would allow for a much better analysis of this situation.

That said, you are going to want to use the output delays of the Behringer to align the phase response of the mids and lows at the desired XO frequency. Then when the drivers have their group delays the same (ie parallel phase traces), you need to then compare their absolute phases. If they are close (<50 deg) apart, then you want to use an XO topology that is in phase through the crossover point, such as LR24. If they are closer to 90deg apart, you want to use a topology (eg 3rd order butterworth) where the drivers are in phase quadrature. This will give you the most symmetric horizontal polar response possible.

For your mid/tweeter section, you have a choice, as all topologies will create symmetrical polar response. You effectively have a large MTM design. If you pick a LR24, you will have a very narrow vertical axis lobe at the XO frequency, and if you pick a BW18, you will have a very wide vertical polar response at the XO. Both of these assume the drivers' group delays have been aligned at the XO frequency.

Assuming that all the drivers could have their respective bandpass output delays aligned so that the drivers were completely in phase, I would try to aim for a LR24 ACOUSTIC (not electric) response for the low/mid XO, and would try both a BW18 and LR24 (again acoustic transfer function) for the mid high XO.

Please be aware that the total phase response of the Mid/high XO will influence the apparent acoustic position of the mid at the mid/low XO point. If you use an LR24 transfer function goal, then the overall response of the system through the mid/high crossover point will be of an allpass filter. and the total phase shift in the stop band below the XO point will be 360degrees.

The first link shows a line array of questionable design. The builder even admits having lobing problems. The second link is for the seminal article about line arrays. It should be studied and followed.

I think you will have serious lobing and comb interference problems with your design in the higher frequencies, especially off axis, leading to lack of highs.

I took a look at the two papers that Dick West posted. The second paper, which I had never seen before, is really quite excellent.

"Line-like" arrays are a really big deal in sound reinforcement these days, and the second paper references all the seminal AES articles on the topic.

The bottom line is this--The vertical directivity of your mids is continually decreasing with frequency, and therefore the radiated power on axis is increasing proportionally. Therefore there will be a natural upward tilt to the response in the absence of shaping filtering.

Also the apparent directivity of the mids will be very narrow vertically at the XO frequency, and the tweeters will be wide by comparison, and then will begin shrinking again with frequency.

There are numerous ways to deal with this, but that is a long post.

As an example of a practical product, take a look at the data sheet for the EAW KF730 (link below). This is a speaker which is designed to be hung vertically a minimum of four cabinets deep:

Notice that the axial frequency response in the data sheet slopes downwards from 5khz to the bottom of the system's response. Above 5khz the axial response is essentially flat. That is because the High frequency horn flares effectively control the vertical directivity of the box above 5khz, making the system's power response (approximately) constant. Below 5khz, the effects of the adjacent cabinets increase the directivity of the system.

By the time you stack 6 or 8 of these on top of each other, the vertical directivity of the entire array is much closer to the sum of the axial directivities of the high frequencies of each of the boxes. Therefore the axial response flattens out, and the power response is consistent and controlled across a wide bandwidth.

Even power response is the name of the game in pro-sound, as you have many speakers that have to play nice together in acoustically challenging environments.

That's a chunk, but I think I can work my way through it. Thank you very much.

I am setting up Speaker Workshop to do basic analysis on my PC. If there are other pieces of equipment that I will need to perform other analyses, then I'll do my best to get them.

GRR, why am I still under probation by the moderators! I'm obviously not pimping webcams or "member" enlargement...

Anyways,

I think I put too much of the gory details in that post, i am sorry. Here is a shorter answer, less detail.

Any time you have multiple sources playing the same frequencies, they are going to interfere. You want this interference to be as benign as possible.

Several things at up to determine how long it takes the sound, at any given frequency, to get into your ears. This can be thought of (weakly) as the group delay.

There is:
1. Propagation delay (distance from you to source b/c speed of sound is finite)
2. The acoustic phase of the speaker. This is basically partly a delay of how sound couples with the driver, and partly delay that is a function of how a speaker is a bandpass filter (ie both a physical highpass and lowpass)
3. Any phase weirdness in your amplfiers
4. The electrical phase response of your DSP crossover
5. The latency through your DSP device due to A/D and D/A conversion.

In the end, the apparent location of where the sound is coming from is always behind the physical position of the driver cone. A good first step, therefore, in designing an active crossover for a system like yours is to use the output delays of your LMH to bring the drivers into phase in the regions where you want your crossover points. Since the drivers themselves have varying phase response with frequency, you have to do this delay alignment in the region of crossover. It is also often prudent to apply eq to the drivers to flatten any response anomalies in the octaves around the XO point.

Now, with that done, all (common) crossovers introduce their own phase response into the mix. So therefore you need to make sure that this phase response, when added on top of the drivers' phase responses, gives the desired result. A good assumption about this result would be that you want the speakers' response to be symmetrical about the horizontal and vertical axis, and ideally consistent (or smoothly changing) with frequency.

There are tons of tradeoffs to be discussed here, and this is already super long, but I hope you can see what I was trying to achieve with the first message.

I think you guys are talking past each other. The original poster had an idea about a "line array" without really understanding what it is. Also, he meant it just for home/studio use, not for a large auditorium.

Go here (be patient as it loads slowly) to see a picture of Jim Griffin and Rick Craig, and their joint project of a line array.